yes! On Wed, Feb 2, 2011 at 1:12 PM, Kamen Petrov <kamen.pet...@gmail.com> wrote: > Ok, that brings some clarification. Thanks. > > So the correct scenario is: > 1) softphone --registered to--> opensips A (pure) > 2) call is relayed from opensips A to opensips B (the B2B one) > 3) the opensips B connects to the termination > 4) the RTP goes between the softphone -> opensips A -> rtpproxy > > Would that work ? :) > > > > > > On 2 February 2011 20:04, Ovidiu Sas <o...@voipembedded.com> wrote: >> >> The nathelper module is performing changes on the received INVITE >> (changing the SDP). >> Those changes are not visible by the b2b module and therefor discarded. >> As a result, the nathelper module (and any module that is changing the >> initial INVITE) doesn't work with the b2b module. >> The only change visible to the b2b module is the RURI. >> >> Regards, >> Ovidiu Sas >> >> On Wed, Feb 2, 2011 at 12:52 PM, Kamen Petrov <kamen.pet...@gmail.com> >> wrote: >> > Hi Ovidu, >> > >> > I do not perform any changes on the received invite. >> > >> > The "top hiding" does it and the problem is.. it does not change only >> > the >> > media IP. Everything else goes OK. >> > >> > Are you saying the "top hiding" does not work properly with the >> > nathelper ? >> > >> > Thanks >> > -- Kamen >> > >> > >> > >> > >> > On 2 February 2011 19:41, Ovidiu Sas <o...@voipembedded.com> wrote: >> >> >> >> The B2B module is operating on the received INVITE. Any changes that >> >> you make to the received INVITE are not visible by the B2B module. >> >> Use a proxy to perform whatever you want to do (rtpproxy, accounting, >> >> etc.) and a separate server only for b2b (top hiding). >> >> >> >> Regards, >> >> Ovidiu Sas >> >> >> >> On Wed, Feb 2, 2011 at 12:11 PM, Kamen Petrov <kamen.pet...@gmail.com> >> >> wrote: >> >> > Hi Guys, >> >> > >> >> > I am testing the following call flow: >> >> > Soft Phone => opensips (configured for B2B) => third party >> >> > termination >> >> > SIP >> >> > proxy >> >> > >> >> > Here is my config: >> >> > >> >> > modparam("b2b_entities", "script_req_route", "b2b_request") >> >> > modparam("b2b_entities", "script_reply_route", "b2b_reply") >> >> > >> >> > >> >> > >> >> > local_route { >> >> > xlog("================LOCAL_ROUTE ($rm - >> >> > $rr)============\n"); >> >> > setflag(22); >> >> > if (is_method("INVITE")) { >> >> > engage_rtp_proxy("e","<OPENSIPS_IP>"); >> >> > exit; >> >> > } >> >> > else if (is_method("BYE") ) { >> >> > xlog("================BYE============\n"); >> >> > } >> >> > } >> >> > >> >> > >> >> > route[b2b_request] { >> >> > $avp(s:source_ip_address) := $si; >> >> > perl_exec("messagedump_route", "messages"); >> >> > xlog("b2b_request ($ci) ($rm - $rr)\n"); >> >> > } >> >> > >> >> > >> >> > route[b2b_reply] { >> >> > $avp(s:source_ip_address) := $si; >> >> > perl_exec("messagedump_reply", "messages"); >> >> > xlog("b2b_reply ($ci) - $rm - $rr\n"); >> >> > } >> >> > >> >> > >> >> > route{ >> >> > ... >> >> > if (is_method("INVITE") && >> >> > perl_exec("check_for_forwarding_number")) >> >> > { >> >> > engage_rtp_proxy("e","<OPENSIPS_IP>"); >> >> > xlog("LOG: INVITE AUTHENTICATED TO: $avp(s:uid) ; FWD >> >> > TO: >> >> > $avp(s:fwd_ip)\n"); >> >> > setflag(1); # do accounting >> >> > xlog("L_ERR", "LOG: to uri=[$tu] >> >> > [$avp(s:sip_proxy_ip)]\n"); >> >> > >> >> > b2b_init_request("top hiding"); >> >> > exit; >> >> > }; >> >> > >> >> > ... >> >> > } >> >> > >> >> > >> >> > What happens is: >> >> > - INVITE from the soft phone to the opensips >> >> > - catched by the B2B and relayed to the third party SIP proxy + >> >> > trying >> >> > returned to the soft phone >> >> > - "Session Progress" received from the third party SIP proxy -> >> >> > opensips >> >> > -> >> >> > my soft phone >> >> > >> >> > At that stage, here is what I have on the soft phone log: >> >> > 18:56:50 UDP Packet Received from <OPENSIPS_IP>:5060 >> >> > <<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<<< >> >> > SIP/2.0 183 Session Progress >> >> > Via: SIP/2.0/UDP 192.168.1.2:5070;rport=5070;branch=z9hG4bK673604 >> >> > To: <sip:359883409291@<OPENSIPS_DOMAIN>:5060>;tag=B2B.113.667 >> >> > From: "359883327749" >> >> > <sip:359883327749@<OPENSIPS_DOMAIN>:5060>;tag=1040 >> >> > Call-ID: 1296636915-3604-SALASWORK@192.168.1.2 >> >> > CSeq: 361 INVITE >> >> > Content-Type: application/sdp >> >> > Contact: <sip:<OPENSIPS_IP>:5060;transport=udp> >> >> > Server: OpenSIPS (1.6.3-notls (x86_64/linux)) >> >> > Content-Length: 184 >> >> > >> >> > v=0 >> >> > o=SBCSIPUAS 900116523 1 IN IP4 <THIRD_PARTY_SIP_PROXY_IP> >> >> > s=SBCSIPUAS SIP STACK v1.0 >> >> > c=IN IP4 <THIRD_PARTY_SIP_PROXY_IP> >> >> > t=0 0 >> >> > m=audio 17900 RTP/AVP 0 >> >> > a=rtpmap:0 PCMU/8000 >> >> > a=sendrecv >> >> > a=maxptime:30 >> >> > >> >> > >> >> > As can be seen, the media IP is not rewritten by the opensips and the >> >> > IP >> >> > passed to my soft phone is the IP of the termination IP for the >> >> > opensips >> >> > (i.e. the third party SIP proxy IP). Because of that, my soft phone >> >> > starts >> >> > the RTP directly to my provider instead trough the RTP proxy that is >> >> > attached to the opensips. >> >> > Just to clarify, the media IP of my soft phone is not passed to my >> >> > provider >> >> > - that case is handled good. >> >> > >> >> > >> >> > Any idea what is missing ? >> >> > >> >> > Thanks in advance. >> >> > -- Kamen >> >> > >> >> > _______________________________________________ >> >> > Users mailing list >> >> > Users@lists.opensips.org >> >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> >> > >> >> > >> >> >> >> _______________________________________________ >> >> Users mailing list >> >> Users@lists.opensips.org >> >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> > >> > _______________________________________________ >> > Users mailing list >> > Users@lists.opensips.org >> > http://lists.opensips.org/cgi-bin/mailman/listinfo/users >> > >> > >> >> _______________________________________________ >> Users mailing list >> Users@lists.opensips.org >> http://lists.opensips.org/cgi-bin/mailman/listinfo/users > > > _______________________________________________ > Users mailing list > Users@lists.opensips.org > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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